In recent years, a variety of audio signal compression/encoding and expansion/decoding methods have been developed. MPEG-2 Advanced Audio Coding (hereinafter referred to as “MPEG-2 AAC” or “AAC”) is one of such methods. (See “IS 13818-7 (MPEG-2 Advanced Audio Coding, AAC)” written by M. Bosi, et al., April, 1997.)
FIG. 1 is a block diagram showing a functional structure of an encoding device and a decoding device according to the conventional AAC method.
The encoding device 1000 is a device that compresses and encodes an input audio signal based on AAC encoding method, and includes an A/D converter 1050, an audio data input unit 1100, a transforming unit 1200, a quantizing unit 1400, an encoding unit 1500 and a stream output unit 1900.
The A/D converter 1050 samples an input signal at a sampling frequency of 22.05 kHz, for instance, and converts the analog audio signal into a digital audio data string. Every time the audio input unit 1100 reads 1,024 samples of the audio data string of the input signal (these 1,024 samples are called a “frame” hereinafter), it splits the audio data string into 2,048 samples of data with two sets of a half of the samples for the frame (512) obtained before and after the frame being overlapped.
The transforming unit 1200 performs Modified Discrete Cosine Transform (MDCT) on the data of 2,048 samples in the time domain split by the audio data input unit 1100 into spectral data in the frequency domain. The 1,024 samples of spectral data, a half of the spectral data obtained by the transformation, represent the reproduction bandwidth of 11.025 kHz or less, and are divided into a plurality of groups. Each of the groups is set so as to include one or more samples of spectral data. Also, each of the groups simulates a critical band of human hearing, and is called a “scale factor band”.
The quantizing unit 1400 quantizes the spectral data in the scale factor band produced from the transforming unit 1200 into a predetermined number of bits using one normalizing factor for every scale factor band. This normalizing factor is called a “scale factor”. Also, the result of quantizing each spectral data with each scale factor is called a “quantized value”. The encoding unit 1500 encodes the data quantized by the quantizing unit 1400, that is, each scale factor, and the spectral data quantized using the scale factor, in accordance with Huffman coding.
The stream output unit 1900 transforms the encoding signal produced from the encoding unit 1500 into an AAC bit stream format and outputs it. The bit stream outputted from the encoding device 1000 is transmitted to the encoding device 2000 via a transmission medium or a recording medium.
The encoding device 2000 is a device that decodes the bit stream encoded by the encoding device 1000, and includes a stream input unit 2100, a decoding unit 2200, a dequantizing unit 2300, an inverse-transforming unit 2800, an audio data output unit 2900 and a D/A converter 2950.
The stream input unit 2100 receives the bit stream encoded by the encoding device 1000 via a transmission medium or via a recording medium, and reads out the encoded signal from the received bit stream. The decoding unit 2200 then decodes the Huffman-coded signal to produce quantized data.
The dequantizing unit 2300 dequantizes the quantized data decoded by the decoding unit 2200 using a scale factor. The inverse-transforming unit 2800 performs Inverse Modified Discrete Cosine Transform (IMDCT) on the 1,024 samples of spectral data in the frequency domain produced by the dequantizing unit 2300 into the audio data of 1,024 samples in the time domain. The audio data output unit 2900 combines the audio data of 1,024 samples in the time domain produced by the inverse-transforming unit 2800 in sequence, and outputs the sets of audio data of 1,024 samples in the temporal order one by one. The D/A converter 2950 converts the digital audio data into the analog audio signal at a sampling frequency of 22.05 kHz.
In the above-mentioned encoding device 1000 and the decoding device 2000 according to the conventional AAC standard, each sample data can be compressed to 1 bit or less. In addition, since the spectral data of 1,024 samples in the lower frequency band which represents a reproduction bandwidth of 11.025 kHz or less, a half of the sampling frequency, with higher priority for hearing, are encoded, the audio signal can be reproduced in relatively high quality.
However, in the encoding device 1000 and decoding device 2000 according to the conventional AAC method (Related Art 1), the spectral data to be encoded include no data of the bandwidth over 11.025 kHz because the sampling frequency is 22.05 kHz. Therefore, there is a problem that the request for hearing higher quality sound including the bandwidth over 11.025 kHz cannot be satisfied.
In order to solve this problem, it is considered to raise the sampling frequency applied to the A/D converter 1050 of the encoding device 1000 and the D/A converter 2950 of the decoding device 2000 in FIG. 1 to the double of 22.05, that is, 44.1 kHz (Related Art 2).
However, if the sampling frequency is 44.1 kHz, the spectral data of 512 samples in the higher frequency band over 11.025 kHz can be encoded while keeping a compression ratio, but the spectral data in the lower frequency band with higher priority for hearing is reduced in half, that is, 512 samples. In other words, the sampling frequency and the number of spectral data in the lower frequency is in a trade-off relationship, and both of them cannot be raised at the same time. Therefore, there occurs another problem that the sound quality is deteriorated as a whole.
This kind of problem occurs in the encoding device and the decoding device according to other methods (MP3, AC3, etc., for instance).
The present invention is designed to solve the above-mentioned problems, and the object of the present invention is to provide an encoding device and a decoding device that can realize reproduction of high-quality sound without substantially increasing data amount after encoding.